When a telephone call is initiated between a caller device and a callee device, a ringing signal is provided from the caller device to the callee device. During the ringing signal, based on a given telephony standard, recommendation or protocol, such as Signaling System No. 7 (SS7), SIP (Session Initiation Protocol), H.323, H.248, 3G-324M, CableLabs NCS (Network-Based Call Signaling Protocol), 3G/UMTS and 2G/GSM, the telephony standard, recommendation or protocol may provide for the callee device to identify the caller device only by the Caller Line Identification (CLI) of the caller device. Herein, the CLI is defined as the caller identification number in legacy telephony systems (for example, an E.164 number) or as the caller device address universal resource identifier (herein abbreviated URI) in IP-based communication systems (for example, as in SIP and H.323). It is known in the art of communication devices that the callee device may store the caller device CLI as well as additional identification information associated with the CLI, thereby providing specific telephone call functions. Examples of such communication devices include mobile phones, landline phones, IP phones, smartphones, video conferencing systems, IPTV set-top boxes and other devices that support one or more telephony standards. Examples of such specific telephone call functions include storing an image or a business card of the caller in the callee device as well as associations between the image or the business card and the caller device CLI.
The caller device CLI and the additional identification information may be stored on a network switching fabric (for example, a dedicated network database, a router or a softswitch) with which the callee device communicates with. When a caller device dials the callee device, a telephony switching fabric server uses the caller device CLI to retrieve information associated with the caller device from the network switching fabric. The retrieved information can be provided to the callee device using push technology, such as SMS, MMS and push-to-talk operator services. The information is then presented to the callee device.
U.S. Pat. No. 7,174,163 to Aksu et al., entitled “Method and Apparatus for Providing Images for Caller Identification Over a Mobile Network” is directed to a mobile network, which includes a Caller Line Picture Identification (PCLI) information subsystem connected to a Home Location Register (HLR), to a Visitor Location Register (VLR) and to a Short Message Service Center (SMSC). The PCLI sub-system include a PCLI manager and a PCLI database. The mobile station initiates a call. The call is passes to a first mobile switching center. The first mobile switching center queries the HLR or the VLR. The query and the response thereto is either passed through the PCLI manager or monitored by the PCLI manager. The PCLI manager determines if the caller is a subscriber of a PCLI service. Upon receiving validation, a first base station sends an ISDN setup message to the first mobile switching center, and the ISDN setup message is detected by the PCLI manager. If the caller is a PCLI subscriber, the PCLI manager instructs the first mobile switching center to hold or suspend the call.
While the call is suspended, the PCLI manager queries the PCLI database to map the CLI information for the mobile station user to the PCLI information (i.e., a picture of the caller) contained in the PCLI database of the user. The CLI information used in the mapping process is obtained from the first mobile switching station, which received the CLI information from the HLR or the VLR. The PCLI manager forwards the PCLI information to the SMSC. The SMSC uses the Short Message Service Transport Protocol (SMTP) to send the PCLI information to a second handset. The SMTP delivers the PCLI information to the handset using one single short message or multiple short messages depending on the amount of information to be sent. Enhanced Messaging Service (EMS) may also be used. The PCLI information is then stored in the memory of the second handset. Next, the PCLI manager instructs the first message service center to remove the hold or suspension on the call and continue with the call setup. Once the suspension is lifted, and the call setup is completed, the stored PCLI information in the second mobile handset is retrieved and displayed.
U.S. Pat. No. 7,342,917 to Mohan et al., entitled “Multimedia Personalized Call Management” is directed to methods and systems that assist either a calling party or a called party or both in managing a call based on multimedia data. The system of Mohan includes a multimedia personal call management (MPCM) subsystem. The MPCM subsystem is coupled with gateways, which include a mobile switch, a digital switch, and a soft switch. The MPCM subsystem includes a server. The MPCM subsystem further includes a database and a multimedia messaging service/short messaging service/WAP (Wireless Application Protocol) gateway.
In one embodiment, the MPCM subsystem requires the caller to subscribe to the service. In this embodiment a caller calls a number using a device (for example, a mobile handset). An authentication center authenticates the caller. If the caller is not a registered subscriber the authentication center routes the call to a switch. If the authentication center authenticates the caller, then a connection is established between the caller and the MPCM subsystem. The MPCM subsystem checks the profile of the caller. The MPCM subsystem then sends a menu of choices to the caller based on the subscribed service profile of the caller. The caller selects a media clip (for example, audio data, video data or non-medium specific data) to be transmitted to the called party. Furthermore, the MPCM subsystem checks the capability of the device of the called party. The MPCM subsystem converts the contents of the media clip to the necessary format based on the capability of the device of the called party. The MPCM subsystem provides the multimedia message to the called party.
Another example of a specific telephone call function includes connecting a public switched telephone network (herein abbreviated PSTN) caller device to an IP-based callee device using voice over IP technology (herein abbreviated VoIP). VoIP is not limited to just the transfer of voice signals over IP networks but can also include video calls, IP-based web conferences using video technology, webinars, data sharing and the like (i.e. IP conversational services). Collectively, these transfers of data can be referred to as IP conversation sessions, wherein data, such as voice data, video data or other types of data, is transferred from one party to another over an IP network, such as the Internet. In such a situation, the caller device may use a particular telephony protocol whereas the IP-based callee device may use another telephony protocol, such as an IP telephony protocol.
Systems and methods which enable such a telephone call function are known in the art. U.S. Patent Application Publication No. 2006/0034261 to Benveniste, entitled “Complementary VoIP service,” is directed towards a method and system for enabling a voice call to be initiated on a PSTN and directed to a VoIP telephone on the Internet without the association of a unique telephone number to the VoIP telephone. A user who has a PSTN wireline telephone or a PSTN cell phone associates his or her VoIP telephone with the telephone number of the PSTN wireline telephone, the PSTN cell phone or with both numbers. When a call is placed to the user's PSTN wireline telephone, the PSTN first attempts to set up the call with the user's PSTN wireline telephone. When the user can answer his or her PSTN wireline telephone, the switching network establishes the call with the PSTN wireline telephone. However, if the user cannot answer his or her PSTN wireline telephone, the switching network forwards the call to a PSTN/VoIP gateway. The PSTN/VoIP gateway uses the telephone number of the user's PSTN wireline telephone to find the current IP address of the user's VoIP terminal. Once the PSTN/VoIP gateway has the current IP address of the VoIP telephone, the PSTN/VoIP gateway then attempts to establish the call with the VoIP telephone.
U.S. Patent Application Publication No. 2006/0251053 to Croak et al., entitled “Method and apparatus for routing calls to an alternative endpoint during network disruptions,” is directed towards a method and apparatus for routing calls to an alternative endpoint during network disruptions in packet networks. According to the method, calls destined for a terminating point on a packet network, for example a VoIP network that is experiencing a service disruption, can be forwarded by the network to another endpoint. The method of Croak enables subscribers to register an alternative number, such as a cell phone number, a relative's phone number or a work number, that the network can use to forward calls to in the event of a service disruption. The provider can even use an alternative transport network, such as a PSTN, to forward these calls until the VoIP network service is restored.
U.S. Patent Application Publication No. 2008/0137642 to Teodosiu et al., entitled “Mobile device call to computing device,” is directed towards a system and method for enabling a user to using a mobile device to call a contact and establish an VoIP call based on the contact. The contact is logged into a communication service through a computer application. According to the system of Teodosiu, a user selects a contact through a page displayed on a mobile device, where the contact may be an e-mail address, a messaging username or some other contact other than a phone number. Once selected, a call registration record with the contact data is generated at a network server. The mobile device then places a call to a VoIP system. The VoIP system receives the call, retrieves the call registration record and establishes an audio connection between the cell phone and the computer application through which the contact is logged into the communication service. The audio connection is a hybrid connection consisting of a mobile device voice connection between the caller's cell phone and a gateway system, and a VoIP connection between a computer and the gateway system.
U.S. Patent Application Publication No. 2009/0210536 to Allen et al., entitled “Methods and systems for facilitating transfer of sessions between user devices” is directed towards a method and system for transferring an active IP-based session from a first device to a second device associated with the same user. A network server is configured to enable the switching or swapping of an active session from one device to another device, where both devices are associated with a common user address. The switching or swapping is implemented with no or minimal effect on the active session or awareness of the remote party. The device switch may be performed in relation to any active IP-based session, including VoIP, video conferencing or other IP-based media sessions.
A further example of a specific telephone call function includes integrating a visual interactive voice response (herein abbreviated IVR) system with a telephone. Systems and methods performing such telephone call functions are known in the art. U.S. Pat. No. 6,920,425 to Will et al., entitled “Visual interactive response system and method translated from interactive voice response for telephone utility” is directed towards a system and method for translating a script for an interactive voice response system to a script for a visual interactive response system. The visual interactive response system executes the translated visual-based script when a caller calls the visual interactive response system using a display telephone. The visual interactive response system then transmits a visual menu to the display telephone to allow the caller to select a desired response. The selected response is subsequently sent back to the visual interactive response system for processing. The various scripts may be defined in appropriate markup languages. The translation system includes a parser for extracting command structures from the voice-based script, a visual-based structure generator for generating a corresponding command structure for the visual-based script and a text prompt combiner for incorporating text translated from voice prompts into the command structure generated by the structure generator. The translation system also includes an automatic speech recognition routine for automatically converting voice prompts into translated text and an editor for editing the visual-based script.
U.S. Pat. No. 7,054,939 to Koch et al., entitled “Simultaneous visual and telephonic access to interactive information delivery” is directed towards a system for enhancing a conventional interactive voice response (IVR) system by enabling the simultaneous delivery of visual information that corresponds to the voice-based information that is delivered over the telephone. A caller using the system of Koch contacts an IVR service provider by telephone and is provided with the option of a visual IVR (VIVR) session rather than just a conventional voice-only IVR session. The caller will only be provided the VIVR session option after it is determined that the caller has an existing Internet connection that will support a VIVR session. The determination can be made via a session ID database. The VIVR session provides visual information to the caller in the form of web pages delivered over an Internet connection and also provides audible message information over a conventional voice telephone connection. The caller may provide instructions to a VIVR server of the IVR service provider either over the telephone, for example by using voice commands or DTMF key code commands, or over a networking device attached to the Internet connection, for example by selecting a hyperlink. The VIVR server will respond to instructions received by either the telephone connection or the Internet connection and will modify the delivery of the voice-based and visual-based information accordingly.
U.S. Pat. No. 7,142,661 to Erhart et al., entitled “Method and apparatus for interactive voice processing with visual monitoring” is directed towards a system for providing a visual interface to an interactive voice response (IVR) system. The visual interface provides a visual interpretation of a running IVR application that allows an interaction between a caller and an IVR system to be monitored. Audio communication from a caller is processed in a conventional manner in accordance with an IVR script having a plurality of commands. A visual representation of the audio communication can then be presented to an agent on the side of the IVR system based on the IVR script. In one embodiment of the system of Erhart, the commands in an IVR script are mapped to a visual representation. As a caller speaks with the IVR system, one or more fields in the visual representation can be filled in with the utterances of the caller. Optionally, the agent can review or update a field in the visual representation that has been filled in with an utterance. This monitoring feature allows an agent to alter a flow of the IVR script or to intervene in the audio communication.
U.S. Pat. No. 6,425,131 to Crandall et al., entitled “Method and apparatus for internet co-browsing over cable television and controlled through computer telephony” is directed towards a system in which a sender can direct information, such as an audio-visual signal to a particular recipient's audio-visual display device, such as a cable television set. In this respect, information can be shared between the sender and the recipient. In one embodiment of the system of Crandall, a calling party originates a telephone call and associates the telephone call with audio-visual information that exists on the calling party's personal computer or on an Internet server. The called party answers the call and can tune an associated cable television to an appropriate channel in order to view the audio-visual information. The caller can modify the audio-visual information during the call. In an alternative embodiment of the system of Crandall, a called party, such as a representative at a customer service center or an interactive voice response unit, can associate audio-visual information with the call such that the calling party can see the data on the appropriate television channel.
U.S. Pat. No. 7,254,227 to Mumick et al., entitled “Signal-based session management for telephonic applications” is directed towards a system and method for a non-human (i.e., machine) party in a conversation carried over a voice circuit to respond to the fact that the voice circuit has been placed on hold. In a regular telephone network, placing a voice circuit on hold generates a signal. In the system of Mumick, the non-human party, such as an application, may receive the signal and take appropriate action. For example, if the application is in the middle of rendering a voice menu, it may pause the rendering until the voice circuit is no longer on hold. If the application is waiting for a user response with a pre-determined timeout, the timeout may be tolled during the time that the circuit is on hold. As another example, the application may switch to a non-voice mode of interaction with the user, for example by rendering a menu as data on the user's handset in a visual mode.